Bridges have two new optional properties, a creator and a name.
Certain consumers of bridges will automatically provide bridges that
they create with these properties. Examples include app_bridgewait,
res_parking, app_confbridge, and app_agent_pool. In addition, a name
may now be provided as an argument to the POST function for creating
new bridges via ARI.
(closes issue AFS-47)
Review: https://reviewboard.asterisk.org/r/3070/
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Framehooks can be used in a reactive manner to execute specific logic
when a frame is received with a certain type and payload. Since it is
possible for framehooks to provide frames it was possible for this
reactive framehook to be unaware of frames it is looking for.
This change makes it so that when framehooks return a modified frame
the code will now re-iterate (from the beginning) and call any
previous framehooks that have not provided a modified frame themselves.
Review: https://reviewboard.asterisk.org/r/3046/
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This patch allows individual dialplan functions to be marked as
'dangerous', to inhibit their execution from external sources.
A 'dangerous' function is one which results in a privilege escalation.
For example, if one were to read the channel variable SHELL(rm -rf /)
Bad Things(TM) could happen; even if the external source has only read
permissions.
Execution from external sources may be enabled by setting
'live_dangerously' to 'yes' in the [options] section of asterisk.conf.
Although doing so is not recommended.
Also, the ABI was changed to something more reasonable, since Asterisk
12 does not yet have a public release.
(closes issue ASTERISK-22905)
Review: http://reviewboard.digium.internal/r/432/
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This change adds an event for when an originated call is redirected to
another target. This event contains the original channel and the newly
created channel. If a stasis subscription exists on the original originated
channel for a stasis application then a new subscription will also be
created on the stasis application to the redirected channel. This allows
the application to follow the call path completely.
(closes issue ASTERISK-22719)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/3054/
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There were still a few cases in which ATTENDEDTRANSFER and BLINDTRANSFER
wouldn't be set on channels involved with blind and attended transfers.
This would happen with features that were initialized by channel driver
specific mechanisms in multiparty calls. This patch resolves those cases
while attempted to keep the behavior for setting those variables as
consistent as possible.
(closes issue AFS-24)
Review: https://reviewboard.asterisk.org/r/3040/
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The change contains a slightly adjusted patch that was on the issue
(submitted by kmoore). A fix was made by adding in a bridge lock
while calling bridge_start/stop from the framehook callback. Since
the framehook callback is not called from the bridging core the bridge
is not locked, but needs to be before calling bridge_start.
(closes issue ASTERISK-22749)
Reported by: Kinsey Moore
Review: https://reviewboard.asterisk.org/r/3066/
Patches:
lock_inversion.diff uploaded by kmoore (license 6273)
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Added the ability to specify channel variables when creating/originating a
channel in ARI. The variables are sent in the body of the request and should
be formatted as a single level JSON object. No nested objects allowed.
For example: {"variable1": "foo", "variable2": "bar"}.
(closes issue ASTERISK-22872)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3052/
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* The voicemail registration/unregistration functions now take a struct of
callbacks instead of a lengthy parameter list of callbacks.
* The voicemail registration/unregistration functions now prevent a
competing module from interfering with an already registered callback
supplying module.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan
to use the CHANNEL function on a chan_pjsip channel to obtain run-time
information about the channel from the PJSIP channel driver and the PJSIP
stack. This includes:
* RTP information, including source/destination media addresses, whether or
not the media is secure, held, and other properties.
* RTCP information. This includes sets of parseable information, as well as
individual statistic attriutes.
* PJSIP information. This includes URIs, local/remote signalling addresses,
whether or not the signalling is secure, and other properties.
* The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT
function to obtain more detailed endpoint information.
Review: https://reviewboard.asterisk.org/r/3038/
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* Eliminated shadowing of the __ast_sorcery_apply_config() name parameter
causing confusion.
* Fix potential crash from sorcery.conf user input in
__ast_sorcery_apply_config() if the user supplied a malformed config line
that is missing the sorcery object type name.
* Remove redundant test in __ast_sorcery_apply_config(). !config and
config == CONFIGS_STATUS_FILEMISSING are identical.
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The snapshot process for endpoints uses the channel ids present
on the endpoint itself. Without keeping a reference it was possible
for the strings to be freed underneath any consumer of an endpoint
snapshot.
A reference is now held by the snapshot to the channel ids and
released when the snapshot is destroyed.
(issue ASTERISK-22801)
Reported by: Matt Jordan
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* Make ast_sorcery_observer_remove() accept a const callbacks struct.
* Make ast_sorcery_observer_remove() tolerant of the sorcery parameter
being NULL. Now it can be called within a module unload routine if the
sorcery initialization fails.
* Fix ast_sorcery_observer_add() to fail if the container link fails.
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This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such.
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The patch allows ARI to parse request parameters from an incoming JSON
request body, instead of requiring the request to come in as query
parameters (which is just weird for POST and DELETE) or form
parameters (which is okay, but a bit asymmetric given that all of our
responses are JSON).
For any operation that does _not_ have a parameter defined of type
body (i.e. "paramType": "body" in the API declaration), if a request
provides a request body with a Content type of "application/json", the
provided JSON document is parsed and searched for parameters.
The expected fields in the provided JSON document should match the
query parameters defined for the operation. If the parameter has
'allowMultiple' set, then the field in the JSON document may
optionally be an array of values.
(closes issue ASTERISK-22685)
Review: https://reviewboard.asterisk.org/r/2994/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Created a data model and implemented functionality for an ARI device state
resource. The following operations have been added that allow a user to
manipulate an ARI controlled device:
Create/Change the state of an ARI controlled device
PUT /deviceStates/{deviceName}&{deviceState}
Retrieve all ARI controlled devices
GET /deviceStates
Retrieve the current state of a device
GET /deviceStates/{deviceName}
Destroy a device-state controlled by ARI
DELETE /deviceStates/{deviceName}
The ARI controlled device must begin with 'Stasis:'. An example controlled
device name would be Stasis:Example. A 'DeviceStateChanged' event has also
been added so that an application can subscribe and receive device change
events. Any device state, ARI controlled or not, can be subscribed to.
While adding the event, the underlying subscription control mechanism was
refactored so that all current and future resource subscriptions would be
the same. Each event resource must now register itself in order to be able
to properly handle [un]subscribes.
(issue ASTERISK-22838)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3025/
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Created the following AMI commands and corresponding events for res_pjsip:
PJSIPShowEndpoints - Provides a listing of all pjsip endpoints and a few
select attributes on each.
Events:
EndpointList - for each endpoint a few attributes.
EndpointlistComplete - after all endpoints have been listed.
PJSIPShowEndpoint - Provides a detail list of attributes for a specified
endpoint.
Events:
EndpointDetail - attributes on an endpoint.
AorDetail - raised for each AOR on an endpoint.
AuthDetail - raised for each associated inbound and outbound auth
TransportDetail - transport attributes.
IdentifyDetail - attributes for the identify object associated with
the endpoint.
EndpointDetailComplete - last event raised after all detail events.
PJSIPShowRegistrationsInbound - Provides a detail listing of all inbound
registrations.
Events:
InboundRegistrationDetail - inbound registration attributes for each
registration.
InboundRegistrationDetailComplete - raised after all detail records have
been listed.
PJSIPShowRegistrationsOutbound - Provides a detail listing of all outbound
registrations.
Events:
OutboundRegistrationDetail - outbound registration attributes for each
registration.
OutboundRegistrationDetailComplete - raised after all detail records
have been listed.
PJSIPShowSubscriptionsInbound - A detail listing of all inbound subscriptions
and their attributes.
Events:
SubscriptionDetail - on each subscription detailed attributes
SubscriptionDetailComplete - raised after all detail records have
been listed.
PJSIPShowSubscriptionsOutbound - A detail listing of all outboundbound
subscriptions and their attributes.
Events:
SubscriptionDetail - on each subscription detailed attributes
SubscriptionDetailComplete - raised after all detail records have
been listed.
(issue ASTERISK-22609)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2959/
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This change prevents channels used as implementation details from
leaking out to ARI. It does this by preventing creation of JSON blobs
of channel snapshots created from those channels and sanitizing JSON
blobs of bridge snapshots as they are created. This introduces a
framework for excluding information from output targeted at Stasis
applications on a consumer-by-consumer basis using channel sanitization
callbacks which could be extended to bridges or endpoints if necessary.
This prevents unhelpful error messages from being generated by
ast_json_pack.
This also corrects a bug where BridgeCreated events would not be
created.
(closes issue ASTERISK-22744)
Review: https://reviewboard.asterisk.org/r/2987/
Reported by: David M. Lee
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When translating from one format to another it is possible
to inform the translation function that the source frame should
be freed. This was previously done immediately but shortly
afterwards the frame that was freed was accessed and used again.
This change moves code around a bit so that the frame is now
freed after it has been completely used.
(closes issue ASTERISK-22788)
Reported by: Corey Farrell
Patches:
translate-access-after-free-11up.patch uploaded by coreyfarrell (license 5909)
translate-access-after-free-1.8.patch uploaded by coreyfarrell (license 5909)
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This fixes a crash when CELGenUserEvent is called from the dialplan
while CEL is disabled. Currently, CEL does not create its topics and
forwards if it is not enabled and external entities may depend on
these topics blindly since they should always be available. This patch
breaks up route creation and topic/forward creation such that the CEL
topics and forwards will always exist while the router and its
associated routes will be torn down and recreated as necessary.
(closes issue ASTERISK-22799)
Review: https://reviewboard.asterisk.org/r/3010/
Reported by: Matt Jordan
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Similar to how background works, if a say application is called with
this variable set to 'true', 'yes', 'on', etc. then using DTMF while
the say action is in progress will result in the channel jumping to
that extension in the dialplan.
Review: https://reviewboard.asterisk.org/r/3011/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Two variables were being checked for NULLity immediately
after being declared NULL. I moved the NULL check until
after the variables are allocated.
This allows for the "channelvars" option in manager.conf
to work as intended again.
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ARI POST calls only accept parameters via the URL's query string.
While this works, it's atypical for HTTP API's in general, and
specifically frowned upon with RESTful API's.
This patch adds parsing for application/x-www-form-urlencoded request
bodies if they are sent in with the request. Any variables parsed this
way are prepended to the variable list supplied by the query string.
(closes issue ASTERISK-22743)
Review: https://reviewboard.asterisk.org/r/2986/
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Several places in the code were using wait4 while other places were using
waitpid. This change makes all places use waitpid in order to make things
more consistent and since the 'rusage' object passed in/out of wait4 was
never used.
(closes issue ASTERISK-22557)
Reported by: YvesGael
Patches:
asterisk-11.5.1-wait4.patch uploaded by hurdman (license 6537)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When publishing channel snapshots, we currently compute the caller ID name and
number by giving preference first to ani.{name|number}, then to
id.{name|number}. However, when a channel driver (such as chan_sip) updates the
caller ID, it typically only updates the caller ID stored in id.{name|number}.
This means that we are currently giving preference to stale information.
When looking at the rest of the code base, the only other place where we appear
to use this same logic is in app_amd. Everywhere else, we treat the party
information in ani as being separate to the party information in id.
This patch publishes only the caller ID name and number in the snapshot field
for caller_name and caller_num. Note that the information in ANI is still
available in caller_ani.
Review: https://reviewboard.asterisk.org/r/2992/
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Made the vector macro API be more like linked lists.
1) Added a name parameter to ast_vector() to name the vector struct.
2) Made the API take a pointer to the vector struct instead of the struct
itself.
3) Added an element cleanup macro/function parameter when removing an
element from the vector for ast_vector_remove_cmp_unordered() and
ast_vector_remove_elem_unordered().
4) Added ast_vector_get_addr() in case the vector element is not a simple
pointer.
* Converted an inline vector usage in stasis_message_router to use the
vector API. It needed the API improvements so it could be converted.
* Fixed topic reference leak in router_dtor() when the
stasis_message_router is destroyed.
* Fixed deadlock potential in stasis_forward_all() and
stasis_forward_cancel(). Locking two topics at the same time requires
deadlock avoidance.
* Made internal_stasis_subscribe() tolerant of a NULL topic.
* Made stasis_message_router_add(),
stasis_message_router_add_cache_update(), stasis_message_router_remove(),
and stasis_message_router_remove_cache_update() tolerant of a NULL
message_type.
* Promoted a LOG_DEBUG message to LOG_ERROR as intended in
dispatch_message().
Review: https://reviewboard.asterisk.org/r/2903/
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ConfBridge allows custom DTMF menus to be created in the confbridge.conf
file by assigning a DTMF key sequence to a sequence of actions as follows:
DTMF-sequence = action,action...
Unfortunately, the normal config file processing code interprets an
initial '#' character as starting a directive such as #include.
* Add the ability to escape the first non-blank character in a config line
so the '#' character can be used without triggering the directive
processing code.
(closes issue AFS-2)
(closes issue ASTERISK-22478)
Reported by: Nicolas Tanski
Patches:
jira_asterisk_22478_v11.patch (license #5621) patch uploaded by rmudgett (modified)
Review: https://reviewboard.asterisk.org/r/2969/
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* Typedefed and added doxegen for the voicemail callback functions.
* Simplified the prototypes for ast_install_vm_functions() and
ast_install_vm_test_functions() to use the new function typedefs.
* Simplified the voicemail callback function pointer variable declarations
to use the new function typedefs.
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Adds the following AMI events, closely following their CLI counterparts:
BridgeDestroy
BridgeKick
BridgeTechnologyList
BridgeTechnologySuspend
BridgeTechnologyUnsuspend
BridgeDestroy kicks an entire bridge, where BridgeKick kicks just one
channel off the bridge. When kicking a channel, specifying the bridge
also (optional) insures it is not removed from the wrong bridge. The
BridgeTechnology events allow viewing and changing suspension status,
which affects only subsequent not active bridging.
(closes ASTERISK-22356)
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/2973/
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For awhile now, we've noticed continuous integration builds hanging on CentOS 6
64-bit build agents. After resolving a number of problems with symbols, strange
locks, and other shenanigans, the problem has persisted. In all cases, gdb
shows the Asterisk process stuck in loader.c on one of the infinite while loops
that calls dlclose repeatedly until success.
The documentation of dlclose states that it returns 0 on success; any other
value on error. It does not state that repeatedly calling it will eventually
clear those errors. Most likely, the repeated calls to dlclose was to force a
close by exhausting the references on the library; however, that will never
succeed if:
(a) There is some fundamental error at work in the loaded library that
precludes unloading it
(b) Some other loaded module is referencing a symbol in the currently loaded
module
This results in Asterisk sitting forever.
Since we have matching pairs of dlopen/dlclose, this patch opts to only call
dlclose once, and log out as an ERROR if dlclose fails to return success. If
nothing else, this might help to determine why on the CentOS 6 64-bit build agent
things are not closing successfully.
Review: https://reviewboard.asterisk.org/r/2970
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Based on feedback from ipengineer in #asterisk, when the media indexer
cannot access a sound file on the system (or otherwise fails) Asterisk
displays a "Cannot frob file" error but fails to tell you why. This is
especially problematic as the media_indexer failing will rpevent Asterisk
from starting, as it is in the core.
We now display the errno error messages so folks can figure out what they've
done wrong.
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Debug messages aren't free. Even when the debug level is sufficiently low such
that the messages are never evaluated, there is a cost to having to parse
Asterisk logs that contain debug messages that (a) fail to convey sufficient
information or (b) occur so frequently as to be next to meaningless. Based on
having to stare at lots of DEBUG messages, this patch makes the following
changes:
* channel.c: When copying variables from a parent channel to a child channel,
specify the channels involved. Do not log anything for a variable that is not
inherited; the fact that it doesn't have an _ or __ already signifies that it
won't be inherited.
* pbx.c: Specify what function evaluation has occurred that created the result.
* translate.c: Bump up the translator path messages to 10. I've never once had
to use these debug messages, and for each format that is registered (on
startup) and unregistered (on shutdown) the entire f^2 matrix is logged out.
For short tests in the Asterisk Test Suite, this should make finding the
actual test much easier.
* xmldoc.c: The debug message that 'blah' is not found in the tree is expected.
Often, description elements - which are not required - are not provided.
This debug message adds no additional value, as it is not indicative of an
error or helpful in debugging which element did not contain a 'blah' element
as a child. If an element is supposed to contain a child element, then that
XML tree should have failed validation in the first place.
Review: https://reviewboard.asterisk.org/r/2966/
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Surrogate channels would pop up from time to time in dial message handling.
This would cause a WARNING message to appear, indicating that the Surrogate
channel had no CDR. This patch filters out those channels that have the
internal implementation flag set, such that the WARNING message isn't
displayed.
........
Merged revisions 402090 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Rusty Newton
Jonathan Rose
Joshua Colp
David M. Lee
Scott Griepentrog
Richard Mudgett
Kevin Harwell
Matthew Jordan
Mark Michelson
Kinsey Moore